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  <changelog author="Fabio Valentini &lt;decathorpe@gmail.com&gt; - 3.9.0-1" date="1715774400">- Update to version 3.9.0; Fixes RHBZ#2274372</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 3.9.0-2" date="1721131200">- Rebuild for vmaf</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.9.0-3" date="1721217600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.9.0-4" date="1737028800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_42_Mass_Rebuild</changelog>
  <changelog author="Robert-Andr&#xE9; Mauchin &lt;zebob.m@gmail.com&gt; - 3.9.0-5" date="1738238400">- Fix FTBFS with GCC15</changelog>
  <changelog author="Robert-Andr&#xE9; Mauchin &lt;zebob.m@gmail.com&gt; - 3.11.0-1" date="1738756800">- Update to 3.11</changelog>
  <changelog author="Robert-Andr&#xE9; Mauchin &lt;zebob.m@gmail.com&gt; - 3.12.0-1" date="1739275200">- Update to 3.12.0</changelog>
</package>
<package pkgid="0f9177c514edf40d72262b07dfdba68e4c3f8b16d67c94960f0f99db9e2ad596" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="804ce059671303ea9d7a87400586be5d84d0f3fbd41b464bddb9a42dfbd36901" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="51b76c4b0b88c7d6c0c658394189f1c70fff2dfecd4946616a7ddd03a41b8db0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="2669ec9b17e90208b79010dd6a61cb5c0b36a0aed28da9b434d347fe9e060707" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="59b90664e958868fa61aff417879082321d356d43d49c82bd1ee1e251eef8360" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="d836a5a4ab863d3bf10f5deea02d36ed39a98da14f88fdfc4b4fe8a7aa35aeaa" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="05dccc27cd6fdb8b1f45c9dbb9d2536dc7ea71a00813bd497a60c6a18f8bd974" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="1411569a5b349fd440314b2db5230099de02fd182174c6fc2047b94fa6ee60c7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="c113e0f17f92ea00faab819f17a9b9ed18f68cb0b71c04828058ad27c38c3504" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="0336372cc5be35c2f7330d103168302d540618d25eab18b949ed4a0422aff48e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="d3827ace4359d1af50c743584f79b36512661c820f7e8532661463dfabfdc03f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="dbec59bf1b1587c6bf645dbfeb20e53dcf74579142c2db50b1d641332741e09c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="333389204f69bb62f5f6f3a9b8f46b3a481c4276546bcdddac9d99c46c41b050" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="147cb3e9a8e5618b9e966a0e1ecdfeaf28b3f322342e19608ce5914424bed205" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="b3d9bb29bf2453f25222d2fd2705482b862c1957e8a7f3c88f5eb84c455247ad" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="44d052769c507e5a1b5e941b2c56b9675598fc6225d2d30427404b8db856530d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="3eb0e0d7ccfef50a9d4e41f22868b5ea458a49f726ec721c523b96434146c2e2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9cf55e0653c22ddd5a61c48770426e6ac4ec2a9a3a6151f626519f2c9b8b7ca3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="20071640eecbf715fc85ba9333426b3d21a16acab15a15c9ad170caa86b3cf78" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="90c9667aff6b9b397e5bd7305501847fdaa5e4e9b70f0eb8cafa41133c78b0ea" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="de585f2f44edb704d8484364d0a7ae9a483b753425b3c627c3a18e6381b15623" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="a484bd9bce4f8d4b8eb575736ea4bfc6c8d00b8c65295369a63b9bdc99e0e29f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="f178fa930b1dbfe2bc4244de739399897c6dbd12a369efd4a0359b55a4f4da89" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="b44b46be45c8d4d77a4bdd84d313ce35a660bfa06c577be01e4c46090b2513f2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="17111242bbfba8a5cfd415bbe0f8faf0606dbc80eb873ae0b3e13ee071b0a29a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="c2c42ba0771e5c2d22810600b088c0d4974a01f6799a9608b4c0f5d2bc596604" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="f742bbb34c7b5259b9eaef9d38f4e335df5e9db560288868f369b7158866622d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="43df12b4bc97a8d59041fc5a2f8f90d6201b3dce063da0eb12463c7627891be5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="4fab47893508f92b02154f08d75b52c6dfe422f2e192e7cc69067e3af981db6f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="b60b00c7b97cc8317102e3cbccc2e2cea4a4b174a73e5b6aab8101427d5edf71" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="4f43338a27b098655c8da2adbdb24e7c1f1281ace175a1402fdae5b386279a40" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="823279145024b445abe7d230083777f3e8536e1191f0bf7c6fd0790faa76cb63" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="b8f7711d82be04eff9f4d5dc9d098f879d2704dfb590105f926f3c5e7794f50f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="4e4a75214fd8430d023c58cc416ab9a4f9c3c5a0070e640c338f421394662e14" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="42d2226b6025d04b268c7ae39a89adc473d617539527fdd1afd7b4b3ccd727b4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="1e9251e8358c8724d1882e409c759346d5c4647d6e3a61abb936622e50bb2516" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="bdfe36c042c39336170fe4cd348641a3ba952fb913faa1588e703efe7c19cac3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="ddf771c02131c220fe22f5cedafc6fa7b5cd0038d2259c927fbbed870bd359a8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="7ac0757617b8c2c76b05a294a1dd05bc59368307566b0276ef400802b8dd55f0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="f0c6db5366156d5ce79081af5f8d456e57f092e3439a0c8f8e4f89afe13ea39e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="33fcf77d9e53d5ae8306ffe75f58db188089f52f1b68104366808c0f539561f8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="3cd1b59f5c8ffc98c8247589c39fb65bb2cb096495b7cb4f88e801404fffc1cc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="f126fe43b8f979994eb239c0a7cbd5538863b3207f88d692255790da816903d6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="3e381e1595ea2fe7a64628f51efcc0204811524c645ced08d4bef16487d81edb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="17cfb3b453ac01def32dc38cd24921468361aac56ef54dffdcce5ee3c5086d30" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="8ac8df38807d967ad1b07f16ab87c5789d3cd3ba28881bb771757883a619884c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="07358a3bc3234a2f7434dd69a0c7476893b51df589ccc9cc7e18fb6c7849778d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="3960b1d5691ab64746b102c3e8f16f32d2cdd5a0a8d3bac91fd027c0af46f480" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="9ba4d64be7e3c6f0e2a9b8e37b1513efd1f5751c55356ef67ce7e0e9971341c9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6fd79af9b2c0b5b22c9038c4f5cc0620f1d8d6f5a02853a2763a50749cfd1ebb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="6fdaff6ec39fdcc7e6d7ecb53e8560bdf5cf574b36dd4a905a6feaf0b1d344d1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="d5d531d3a91cb02c005c8de65f73ab881cdec2b48d772b99ae103ec6e998fe36" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="23c570ff3d6f191a40ed2b841d4c6a15ebbc4a676366284fb6896d805a25ffaa" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="cf7b62e5c7c3ebf6690efa5c805a2c63782cb949d91f2504e3a1011c410658d0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="ab8814baf5515a00755460a2f7571f40558314e711bce886611848e9ae103711" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="ce77da97c8837f2ab3d9cc670096514b84ffac068a37335663e4afb77c79744f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="99771c53cd126cd28ca4119718af302281bd65c3d6c90353ca0871d360d78f2d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="64c1354a6e37ac3f4eec697795d94c2a5059e7f6b6365a1257a5dfa75ae6ee3d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="bf53d2368d8f14a7e08ff20ebd5726bb76f7de9ae7eff53a9a87da28ff0a9a61" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="1137b3c880125fa78456f8c8223612548ed570d970de7a3d940f3aea1c3b7f75" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="06c0ba6ed97fcb9b3ce994dc73dfd9d48bac0ac0cc2af7b6b0582f5f9b97a515" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="2a21e27a008272eae5b46eb1b1fe71e1da6dadfa57c238f535a708549d6076a6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="62bb3f57fec923063c1f9d66ca6c24d549658b42acf208bfe75c6949fe851b94" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="a02068b3fbac459fcc4c5017e108c01fb46439d7d63102e377fd710169a60aaa" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="c806780775edb45b5ed7049958347e468e463da8041015bdcaf6283a3bfbeaeb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="bdb884dcc4c419df752a552b275bcba1fbe48dc5cad244c12bc195dd6c00e7ef" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="632d24e672f46db3428d058b0c2ebba25b408317879b2f4d37edf269d3752eb9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="feb141d99c8d63cd929d9fe42164511d94ab0a412b06e8ef1f35afc33170d68d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="46b8ae43cf6895fcb6eb549e2a6d7ee7c171edb57654b3a2d6f4233ed48f799e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="9a37aa29c905eeebcb7d4fc13ee2352fc75b15ed5ccc3f2611c0022a03205172" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="dedf2ed1d096d6a4bf41732841cd680fe736a66acf9bd74d03fef5189a2a2e68" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="8f2cd97c4b98d0c2a9120e87ce0a65146bc677a95115304e62716bbbb87a6065" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="514fa114cdb2b0aa4f178c379832be7ef715df6bf864ec4b5fa65e797242d9f3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="5866dca44169c04d938db5a1545a1978867161c2be4d269cd001dfefcc1c9658" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="8daf8ace633787b2a02dae3f58dc81879423af92b68434c88ab68e88e71a8c05" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="3b912ac8a53b022614ecf5c895bcdd509dbbd95be96fa4e6a5d2560c20f4a7c6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="ffefddbe13870abdf6ee6e36f3c4f5a5e149af93e1ed381c749240824bef48af" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="9bfb22fcf0d563e5c2de197e0a8d41df0193d3ea764b6f065b6f517b9da6cb31" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="8719bbb0bb4730f87bacd4b10f17985c76e47b443001be0c61535e2e52acef13" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="973a6098fe94341d2c602a5badb31b95b1989b433ceffb42c5bbe9328ab0e2e6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="147375e1b24feed49df50f09c4ed4d973f42981130bbbc6f121834c123314db2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="bf0e3bfb9f88f6ae050080907f79785acc675a2319546a5bbd4329a1f3c9eaf9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="c0afe2e87259597f9f8564fab054325b27a6b59205c40e1fd650a8b5b7a574d1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="7dbbfed64530ebc8b580b02a3b3ff27f38c27aff20580c88be5f75ca36ecca82" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="5513341df9b473390f0f7c0fa37514d91cfac7eb930ea5427f299c9964c16018" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="edf897dc9a827b3e7feccd8667f1d98e0690405b302cbde038c0364fd2e6c341" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="e980ec70bf4b21e86bdf3759a1dbe44b6c6e274d193d53cb9e7093305283e4f1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="3082894fbf5bc427073e9ba56120c13b7f896f177e4d56e2be4f4b264063cf68" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="3680d7b88ae3cd19272fcff8358fe9bf8f100ae8cc1a55b5b818a8766a5bea8d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="452a213c4b1ce6ff5b0960856163adaeee195c46332a0e0caa0be3292d326c77" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="cba3244196f0772043e48ffb5ea30a1b23f5ec25d49cbecaf0c3849f2cc6a671" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="05042584d82d93a00c3218d55038068d95c431cbba21f5c0529abc7a5cdfdc7f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="9a2cfd3f9c9881c2467da2e0984d0fc1e76f65104b3227859785503d0ffec376" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d71030da1a6e0837f61aa1bc90d6c4a39c7458aedfac24b75144114175f21443" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="27c0c71db2e8c156c13b9ce25b3482dc6ea5eb924f8e9a5d4555e3439983ddeb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="474ce62c20746367d2b9eae8da70cd08edbe050048f8441d01a5647e1e8bc2a1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="c3b39a0c0822b00b7a2738a35dd1a36cbebdf5b752547edabcdddf0a2555b4a6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="dad1a51bc23fabaae35923a9a62d1bd783eb05252c905e34fa7334bada0e47c3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="f8c34ab28704dc098c3121d10b1943f61de0d7ac373501de18e1f741e9122c47" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9074ec094117fe247f37678a2833d062ffefab257db1e10b4c6abd33feec1a6a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="ba45b4dbd7bec842aec4a4894932a5d0bb3f0ac0bd538c3f13fa7f43b8ea300d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="5486b9e75a6c4658bc48d5ced0ed70a39ba4c62bbbe2c21a7a23e0efc5cccc7c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="43e9cb6c3a4728d441338ae2cc47c8641ebbf46565dac08172b149fed291f467" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="8968ad925fa6ffbb0a1bdb07de441a9ef2a9741ec7ae963444f5e37b084da3fa" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="5aa09c97dea41e4c034bc9915529b70c0ee75412721de0b0877e5e0ad195c146" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="0e7abec5bf1eea3bdf766fbd32e1e1f6bb684dc9a7b99b73ec9d5f98e12a62d4" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="32a5870dadbeb8f4c6195a15fd7ee474b5b3a724703f1c72e19c201af798b98b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="6d132c567b4657b07cd5be74a6836e40c66c82d055f72519742fde2b26e4d1cf" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="be7582174e12aded922a9fb35b334db40943df88749cd02db4eb1245f5a090a9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="7ab53d51bb9180d7c1072ce7b20e97042e57c4762974fb11107dcc6442c1d9c9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="d1740557ade0e8af7cfd01dd4c813e0351443cf27d724c76d16bcf1d381de782" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="e5fa3c53a3d5313a583ba0c1c9dfbecc95da151c425b5c01c9036a7efe295356" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="94077421528258b172bf981237585e529d2785caed24c7eeb0d876adf8923570" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="2a594e2e6dc986fd01690e5b8bf65d1cc54e3ee03aa72e08477ace46364a2538" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="af5016faeab25509b47d8f6a11736f654ddf0e14f116a80c2b2eb72147474a89" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="6ec0e9dcca572ac79699b2c8a24653b199b8f2e4e2998074368e25f2e74b4e3b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="9b1ecfbb161e47822a6300ca300615ab6e3c7071078ecc949fd4afc377390ab1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="3f7e45a79c391c482164aff33e5ee077c0ff27b27a65790d11e2747447517902" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="d9fd1aeede3a0a2d2f069de5dd7c155c220b7df2c73166a1f2c38032d18a2a76" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="71d94e862d3c0444bd2367bf224d5e7de2c90c5e233d411f95dbc0e2c41aad89" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="678921b9db8c279ee292f2991693a3e20c9a779fbf61e818cc2c102a3c52c73d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="935dc486c9aa04deb926a1c353e84fb9e1553aaceae88660e609d3e40903c212" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="a3857b05563715fbf67d109fcd95b68b1fb45c7bba0f785ef0c2920043f45455" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="6fd85dc3647bb6db5a7a8a44c64a7df2e2644ba3b985049b50c4a473f32bbda3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="b2f4b5f22eac3384072934e7179d772974d9b646285e32eef15f6fd1a85120ca" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="f80edf4ac1d015764af9ce76279feb546a4c2d7edb732783d6137e016ba76c15" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="26ad9efbb0678a7ae301b6b0c84015534812390e697047c6d4f559ea483c8f92" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="0389b810f846fa62be8dcc763e021c39e1952aee72d8a9bbea5ae6cb54db1657" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="877d968472f3ef661d01ddd313cae96e0d8184be2cb1d9e9707f6b6f780ea768" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="66f5675d6b2aeaae22327d51ea33e305cf5598ddcd9bd05a6c47633424781c9d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="9a191f261e17f74f3efc2d4f61c9aaa8e981872df4f674ac02cad4b015ae7311" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6c260eb80ce663841405f201bbbf522536e10cee7f951f784e2bc9348bdfe2ce" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="dc0320974b42f24139f29c1403e4fa002b3ae229fcdfee0241338bd0302bdf6d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="38384c43b5d4fae3d4e5256192cd8adf23829b1786be7111509763c81570511f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="c8501ec62279120c817bd38d17bf045429d7fe3ae128ab8ea084d0d94ec16859" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="60506f5c2413f4de23d197ca43eac2b64bf3a2498272e1c4b5ae688766893f93" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="8976e713a1c362fa069b99ba0dc5f69fd7ebf7b5b0529f8a1f3f67d0a36c8180" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="057433d5d173ba9349187804e88e778a265b7cd6e7200ac7125c7eb5d33866d6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="800e8b1fb771ca04b63858472b9b5cefc3c29cdab055f5c87fe0efb5057f60d6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="290102984a8acd571fb9002b6fd9325745963af4658d502e782bb3832335c496" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="ef62ad7ab86fdb751e13d0577d714cc56fab776e2522326b4236a358e36fc695" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="f356300cb2a1ded2b2ca798b2fab43033f9c1daf455df7740e74b3228c6f9cbd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="0d7a86fe1368c02dcfe94aa7abd5f08f6f3b3d72f9bbd0cc52874786a298d90e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="d297b01307f2af01df9279fa75645d2a4add47ee063f4b67e6de747d6410e7b1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="fbc3ed03f13140e89bbb5c3418aeed81c7ec11819e3dbff37b3f5aaa42c905b7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="e180669bab176aa774ef449671f67bb1870b5edb4cb879880251b2b454ffcf9d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="5d76d4bce12a6be63ce1ff5d997272de7921f607ed7a1c5c78fe2fbad1f49786" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="e379388417d2ba33678973904430ebc0b723ad303c69bf317cdfc016589d0f41" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="4dd1a4ff03329bb8fed3db25f9ed29e9bef4d03a7e29aadcc22da8997bdbdb0b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="1516b24c998e471f609e0bce7eac80749eebd89ba5a52540303f2ee817ba349d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="c45d0366c6804f03513b61e7c6575312d5ceeff3806f47f3ace44dbfb91938ea" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="ed8633e8bd6174a95476aa72865510965c168685a24350bce9d5456171857c7e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="fbeed3684e7f1048f34417721e348ad1f150b8dcdd011cf8279b58a42075d34e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="ad18c5394efca7dc299569192900dcd17d510fee7fcce1bcc29580edb92dca2f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="23cf875efa51a5d93887af25d513ac95c1728a3cd99b174db4b26cbe3427a676" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="e15197fbea4038d239956ebfa7da7c476326e1d4acff9ff1e04b293a90f3c101" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="bfb9efa9918443c34632c0f67c85bf1e0db29a8906828fbad10e2c404c0d3b45" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="2eb9eb5f8fcfbb6c9b9684a8d375b6fd1fedc868193a305627074159e321bcf7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="15d190e62a0446e18b6d67a59b19dd6b6a1773044cda0d99fb911e7092f5c354" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="cb06056e4d56242f5164ec57bcf3d9b851cd3bda9e85378f806b7e1de5161696" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="ef3581d3c12618b903accb4144b738af002f6449efd723ab2cb3d8e417b6ddbf" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="1ff50429f82856278e35a5834713599a25109c52cb0dfb0e1bf9270d2aac3971" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="62797892a7032b2edd78c1b746a4d3fbbf8a121fc852ca3f0696e88279218a56" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="ca219cfd26cadf369018afbc0dd50446ff501c0b42beed2e522fb8ccb2f48932" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="8551fe9ccc291b72bebdcddda17af0b77deb600381aae63c03ac2971fe3f9c2f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="3ea29c4441da2c28044305acf9acc2057a78d04738d7da82b561912e4dfca4e7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="9bfa7aa86e3481b14f931e313fa6605162f0c233831cc981fbeb621df0119089" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="1e4db7e3fe8441747b56b646649523d0a5a4fab11321d5ee5b4b298d15ae529c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="13e0d40b0fe95e623551b649985bd1d3c59a1f694575186b10e319e4d785db4d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="5519eb556e35c54c647e8faecc165b5f24e191c404a6110ba7532aeaa38f34e5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="f00d1b845c7fe6f3981e0c22bb52a5da11446e10cc6e5f2effced88eef802cdf" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="77505ebfce745cc8de3075576aacb768980036840306aa5f5281b4ff80f102ca" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="36bb83faaa7bbc64b60bd16e7961088d596ee0db7e00735c93ac549481685d54" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="74d7fe0c09d9f3e5ccc4edf506cb5c812128998358b912b59373e8c183de66b7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="fc2d41e67c9e18a75759450d27e7f149cfa7e909c583c7559fc55b647d665c9d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="5b3abc4dddeb4c91c8a291d122927a36d31c293583e94b90b351fe7ed051c1d1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="b770db77e754530113f897636d28fd865242549b8a6a5a0ceaef2a4faa789c96" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="0be8ce860c17fac701b848dc19c821c299d5071f1b2624e85fd294df417eff1e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="d472956872dfa32c4af7418e841235ae5797a2c6f21890bab2a05bd32203e764" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="3104ce7af52342ba06894a79b3eae14c4b4f0938f82a4abf812df64015bdad40" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9fcd5ad5322b05b0372f1ed8b9a5c72e3628ddb89d95f57dba61043630a972a0" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="0e3f0d1557425d308f16d30882f2ada511f5cfe2d2940e9d4afd330cf961ac47" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="14ec34c1413fe1028e191ec04289f227bc7656d050c256ada059fc584e3d01b9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="90d3f49e7ea0cde8e4139d5cbc133b44238320433bfff9e3ed97544fba53431e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="f158b458099bc7d4a4628f5b1e74a85ba3a7c6bc4fed119b713aa0d384b6d5a6" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="ca567ab4093ce48721d1a207caa4a4f3b0fba0d5202eb8f98cea52e94968bd1d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="50cc61fb898814df59b5d1ff5e2cf44d876415565aadefd67a96b5c6b282117e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="addaa6d952052e658d67a4bec79cd2e293d1960406f40c0c4679950292df72e3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="641bdca4599203b3735a435da03820ea9d1f9a5f41534cc5287b6283476dab0c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="ee8cb21ba251f60c4ea19dc7271bfef162e37fe085e621b99b3f5e3620f46ddf" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="087e9bdc4a6f9c88ee89251a55e6e1abb7dcc47da72a76864b2e5e32a14de558" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="ad0d93c8a8f8690712c17ba0ad7ff9467557829774e4ed50e0c3e171e39c5248" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="cb735c83b428408fde4b271db4cef8462994578b01af0e8c547ef535b8fa4533" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="af087682a159c87f5f8b59fa3917f12757a6ac60bca86f886c7978d0dfc1b8ef" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="10fae3f7abd819c61d87061c201c82ff2a992ebc6e0f509581c333620b6cadb4" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="d3b47ae59fe5eee05e3a4a592a7880bb68784ab767e63f9bca866307798a9d12" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="627a049b90bdf15bb8616ebb9dcf33f4f569516e8a82e33a56178905756ce4b3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="81fe42900c8ead93fb644979d2b05a8ac437a82f3cbe45c5d7928f436a30cf3b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="e81edf2fa2bdc24c6d6b65f1520614d14e4502d52c34a22ae5c4f1d8b4ac38fb" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="d97e800add22756ac34d6f7b634de7044ccc14122a5fe979feef46e1aa623e06" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="387ac3febd2d4d819bd3f379f9f9efb3e690875ff18be4f6fead7251ae9e582d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="7e961e246ec0aa6d4ca5488e1a6a78de92c5364acadb5ab840cac70bac47959b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="b41eeea2a09b5c78047610a91ca8d86f85a992951fe7f8037c0daf7bd76d8540" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="d360aaa496bf8932c3e69b015d8caa19d8113c45727bc3d5408e2ae5aa795421" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="8fbe1d375bbbadaf0633dfa3418ca95e6d8ad6597463ef5ff27eda9b41dcc581" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="0a7841517dcc2cc0bfa3fa062fbda7ae1f3eab56c4e9cbc02f4e1f93400e9970" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="701c983fc94a786982e083ee9a70790a1706b8d16173279f0569e262b063d8eb" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="1acc2d145b3a953e470113ead10b544756c5ec1d2f11a6d0b2ae0eea48916838" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="ef70e37dbdf2484ebebc0c67fcaf43d369ce24460cd9480427b82c5bd1b9055b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="3369517b466e3c517f6d2cad046fd86be8aa58cdf104afa7607d061895d6cb06" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="130044fe9d1e12cd84b0ec7a43a85fc0b05ffdcb280915dac1307eeb2a4d55d1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="b80bb562d5e7afac8eda8e00900cd9be9574eeb5c621fb62ece3df3405d7cece" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="42f600fa40e1bac4094f9138d8218b229ff38004a35fe5296a38150ce35fd733" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="e4ab9befb438c3db224d5fceadca608257c0ad6c219f9c1a15ac4f8c60b25548" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="90933c0fa445c105f4996fe261524aacb202e1733fc91946f9b3ff12d4e1035d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="7aefc870d2a6015b175e8e81a863c4590fd7535d5bc3584992006f9c720f0dc5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="6e5ee9d5e07a9cb5b31e4a1c0976e2935201c2b43786b3c5011a08e45f264cd9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="64b70cfda09b13f522ab444e07f8cbe13c49cbd48313fc2bd80545aee81fd5cb" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="31d69879e695e42f7fedb4a19f1aca449eeaafb59663b697e9309aeccfa742bd" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="43f1e5501d3bc0014a1c88e4139104e9875048da30c6bef3e5f1580cd086e04d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="32cfcf5601949e5425a601e09100bfe27405663e246f137c7dc0fc9b12d54599" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="f73ba98c2fb42511d67795d71fb3a5553814f9a038a4a954fba405e32dad17b6" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="49d7cfe2f2f498ee691b0b65c5038fbf28047ea552f4ed2ef8d31810b7924b8e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="a50434987d8bbac3dc77f46eb61e4bf96ac243261e1558285464b4bc7bd48339" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9ca8cf4ac0a9d078332b7635fe3dea912d90bbcec8d0c2aacbaa4911505c65cc" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="005132d45b2fcc87d34eece906592f7d6acaec05131599ef9ec147e2b231c2b3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="42267cd3b78184045900ed919852ddf6931ed9306130625550948cbd3f8dd58c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="4dfe5916b723d3f08d98d04e4cf719ca6177a3bca4272bdf7726ffd032cca9ac" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="9173f07fc4f48c9aef6662a1100ec25263101009d9562cb47d3ffe5dcb6369c1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="2fa3e785df56458178c9b52d07ab5a7224f4060bd80c0f5e7ed1644449becb2a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="65acbbcb57309f044423a0b57a0327e76df3c2dfb52e1c3753a79b961a642f77" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="0f07294507fe2b5aa5a3f2b938c88c8a7dce990b72a0c7fc95dc1c05b0a13ff5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="2a187286a6502edc623f8187d5c6bd5fab4b52925e19b3e31399d5efc329281e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="32b05d5f5f24cf0852622cecf080845c9368a932504d4d75a687c97b2fa579cf" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="b266c7335e7fec300042626ef55e21c1a662bf755a0a33cebcf0ec822024b8f1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="8b9f72e3f865e22de83ed39cd696a43616143f961e499b9123d177ee611a5725" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="def4bf0154e592dcfe7adebf6f9c136e1d36304439fd13611c9f1c0c4279f792" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="0885882aca04c16eb73506aafc7836ee84ed73635d83681398e83c17274f6a31" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="15310fbb75c9f13e3e03c63e05c977bc1ad8a6c07b219686ff26b9d96f48bfca" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="faa74f0f3c2ba674ae763e14e14e5ab14230d4e07e82136728d0af1d333e179a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="d02d98c0c1df734a525f43f7fbf7fb6b90e874bd91413593461db6b0ae8f59d3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="bd52fcf8c30cb210381efb90489d9b243aef77db46d24938548644c74fc69f77" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="007b14ab3649ed1fe8656e7d4ee0635f3dd46437d56e804e153c4ac52a0639dd" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="0bad63056948c4f73b4f4e8b10585bad5d0f5450d4e830fd04618d5d4e071238" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="34cba37720c6d3d4ba3a7bd88631c3cff1ee71dd33576d3858274b6432760999" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="2454a7071f4b829a2aa20377592348796185d3ee8b067b0b8ea040d9ec5f6063" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="7c8c1ee7960880387db173bfd2cb3d153fecdb69bb5764d9eb0ebfdda867958c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="9d4ea19a3fbfc51daef08dfdb6b04ac1c9950d598ae8054e09f6b9868867101a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="9a491096c3d0b21f2e8e7310f0e1dc6e6f30d9398e3756a8ed4679de6fa4e297" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="16f8b8f4b507b9360d943e59a5563e063afcde60857d9f24fdd4684d66e811fb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="9e56135f638429b9c53cb5919c269454a3c0b51940245154f191dcdfe7a8e846" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="36469babed909168c938652a5900c385b5c42dbec945ba11c96f70e2ac63df98" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="126241c66a3095b0ec55bc95372bd411c5b3facb396cf47c15227ae3fc3d971a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="6e7df5900478633a1382a906cea214c284b66e731b83b033b6adcbb8098af723" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="bf0d053b651a3732856eb84fca25616c122db85d827fd430c3b003a1f4e18ebd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="c1502bd23cde5bdf2e9ffed62d73e2af8a7f5076c2d4534f536e8fec1f22817c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="6cb740d5a9ea0a35239a6dd32e28d340a4d13e34ca94376a5b2c684ee4710ca9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="2e1cb2984b228e6ced2f50de4f71746faab99456c4fd4857c2c17e823bfcc258" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="3d5c2e90a212dc33a938394c28953f2790b8f96a88e33acf4ba51894c9a1997b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="13b5ed1e15c37debb6aac6038da50da3c6aa59bd4052a72249828fda81e5f668" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="412a166d70bcbfcdc68593d1faa0add670228a11be5f213ae3d0428b70610a74" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="2440aeb9360316908c6e5b87405f835ab279f9619102193ec925b90543b00977" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="ba81ca59840d37ea64507e34e919f25f11083750e0d579270b754e179ccab74c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="7ccd5a2e922f978be8820d577d5f103e0d19c1e93b45a4d3bbc80c61c9f5edf8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="289b1d1c010b26a6b128941c0e2280d91f1da7567fda91bd51291a4a9a215ab3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="ced4767d8b28109158881e6eacb9f73487a7a4b5e7bafb0590106f0e1330fcba" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="c22aec4eaa3b9ea0359a515a8bf4aa2e43a936a70f144cdc2df0ab007e14e6ab" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="acc85fbcb5d00c9f14d321407e4fa1e0e0304a250467346a898b30b5cea8c27e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="e90fac91fcd44712b8008d1d3b1d682e83f7013eb59640c99ceca008c1d3f8ea" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="5bf768a534bf2e1a7605f3852a4b75e17806b19ed4113a1632a2de42be39dca9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="70477791ee69fa5ed09040d86945d9c86197a113cddb4463b92ab714e7b9aee0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="2a02cafb7efabf65c7bb11be6b6133e2905ca0eb35c29bbfd21dff24a0d26bab" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="47d135cea0070be271b7ebf61918bd0c87101c08f77373de5a7879deb42ea873" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="3c57af715a4aaf8bc1367848119860868f9407a289f3af327e0aecd05fa49ce4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="a525021ac9a332bebef5ce533ed18c32906935eb3e29dd31456a935c80d45ab9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="ffbfa50ac46bcdcec68c92ef4c23d6fec9770b2d96ef09cff4e94bcec1e1231b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="2fdb7ce66a98fb068d195b4a8ea19c8a08ab12298610df4bba739a14e1851b67" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="7d15d914351db32da1396c9f8cdce5d3e666c792af14a3567153ed2e3dbd1de5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="b72f9df25a3ebe431fa214afe93d5ad1d2c3cbdb450181f9cb486bb1535bbd47" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="d8782aea395a76a63c8c6055a1f8e9c05cf8c41a4720daaa14874b6728d5f1ff" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="81279f4fecf97f8e52b7a92cb1108cf85f15958cf9a93d4c8f4a4a910a3b47bf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="eae2cd46ba34fdb236a3a8043583adf9f2a0606cdc4386f64dd7947f2991e028" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="5e71de6a260fc91cc97f5a91cd9de796f326e5acd0365adfc74289f2ed0fbffc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="66431b1d4dab12a64d005e0fbffbb5a0fc73af6c13b365285036965cd10d56bf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="c27060307efc9ed2bf9270bc86a30f0a311b45d3e007dbc5f0a689027757f0fa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="6e8c6ea0544eda8b67eae371223c6e55030f5074034f5863700c373eb5f7ac50" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="7425e630c1177e665fe4c9cd71e78c06eaf2d290802edb87f455e2660fffe04e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="09e51d8b9fb0391a0dd848f84ccd3043e3af919cb1122f3103da298b83cad86d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="e18c765a90ed910a8bfb3b1caaf66ed0897df595ff6ed2d3b89df14ab9d3e739" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="36ec2f4a6635ce654d976fa46665792f97e6f486e636fb404ee695de22ed93e9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="546ebdaeab240b28ba68a16afdc73cd616d0a96380aecb3bdbd13fbafeeceac3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="5e3f72a044fd1804f6d7330d0282212bad0950a04b0f98018437601f751525bb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="6efe46d62c874c386a6bd93c42ce5b1577abba9e761c0306f24f0f262fcb49b7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="8a54b52426f5d811e7dd0653c70e4f51e3b9f5fd60b2846627d32cf50c33e62f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="4986367e6ab879d44351f576cab8366eb791c82ffebdb937ad5a79e22866a06d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="39c641ec1b2f64214bff3899b961050634878ec5c75581bea33ef84f44fa4125" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="920e517a14110d7cebf5dd0902157c6cfcf21acf3e78fb25efaf522a16d789be" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="ca181527953387713a5fa3bba4b70c9ca3023c2adbb0a5b129fb1f82fa0c9823" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="82eee30ec5edd232c53a44c0051035805a6b93afaa51f0a46ad466174d41955c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="9042ee16d84c7f92547b0fd2cc934d30e551348c142b3aa90d6f00be08d79ef4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="725be3b26ff8ae5333ec4d69869e9bc9ae405da7e5b6cfd99a62efde5224224c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="85dbb59061d9d0b50d5b97513d00ede30c16d5e9214658acc012c9ce271e53b0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="5504598d06b233401fca5c0e447a2988060cde52d24b1ac4ec7260b5e5eaab0b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="5d896d165367a2cb4795caacbe3b18a7fd7f0a8c214630880c3cf7b767c80325" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="d57d9aabaa9e7fd1e4aa4505b8659fd5c515600e6144106d086fd8d8f0c905b5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="d0a57a119ec49d2888b16081594e03c6d712c98f3003637c06c199730b770175" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="96966932a5194e819108791c23d05018333ba00991ae2a1cd14a450578b7d9a3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="9ec0a534733a1ca367d2314b9024d8267b9af5a0ecd6693fe23a9b999962b330" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="82eccee66112bf5851ec0ec9cbdc44d0bc3335ae19fab485a692bf6dae2a27d9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="6561d10ccf57ba3973e0b2c7ffcd1516234fb2eea80824be87c860d5edefc0eb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="de2d025494a98249f54711a9cd091c4ca8d114a20912e7b80f6b70f16690f199" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="562b8e4a2a158bbc1e6c6ef21b6f92d72862a63127ee299a716e83c808ab7ce0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="66d78f3686ce08f0967ae8f260b8ea5e1d4f4065e9d1817b47be9f046a30c06b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="97fa391bf46891e4d67e3cd25817430ada778108290229679a77e11444e678af" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="a213636f117401d84bd4207148b8975489b60cb0c37d1b88d3dff6cb8497277c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d6f30f46e99751a9e74221b5f897011b74b1bce1f9d475dde6778766146fe848" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="b750b5d54319ea4e8e53985e2745e20dbb0e470ee10d4fede20668cd3705474d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="c21dcec5ae5ad3718d3a0220055a062b3467b3006ca1fabe621d110d00620179" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="a4d8b50363e4e0ec8f769a8f47f204b5db9a3c953c4b1c7c105c98f54fc1e399" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="1ccdda37b56d61873a3b4e35f18fe2ae61b60ec9facb434b845a4b0fdd94998e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9eee6ff29646cca7f8a76ac0c506e8e8ba0a979bfd6f03cfcf3aff2484ee6c87" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="6ba1a61ded1d0795576708c81525d3558af84d07ce15b8dfe305398bef530321" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="bf570098a5e372c9afb299a17821169396199d6ed393d6071b77799c537502c1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="cc500a8c1591bcf2bc7c8eae95ebe2176753b2a649b628a4e7f84664dd3d4c7b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="914ed2862007c51b0b0c102d9908415c3888802f828011fe931c49022747e13c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="2a69ee83b3d8b8f2421b41163d03eed7e4e4b3acf62a51662abf7f559cf9f0a4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="a99897304fe960040eacc3242b884eb8aa8978a17b865e4c0f0b65fc3d9610e3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="0194825bfe7247321e54fdd2d3c2939c825dfad73a1f150af4c77bbed4ee0ca2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="233e5a6a0dc66ae49b27d707e509c224b032055475931fddf45812b17386c8b3" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="e83e96a2a2046b6f2d6cf58c623c0eb4aba7a2c5a343f3848796a306afd0f8c8" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="3a0570d172d92d6227badb10cd2c186b65013330d817524fb648130cbf499891" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="e00fff7dc81bd708d4b1f66873573748c36d7fba1f545ff0c22c196d222febdc" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="7fab8d9ad1d74382d3b92ec69be9c00a751dbc74c86990a92f189502c1431d7a" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="6b85b72359c14897d5661628157eef7005cd733be5758855f18cd846e38b1729" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="68e21bbb57187e56d02d1342d825db1a24856c90a4e83bd5fddc84c588715baa" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="81439e5ad9f3c436cdef029a9d50f753b3946ea5645ab495658c88e9a419eaf0" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="4b91c62c74ac817eed5d2714ef12bc4d047e90ed3ec1f28e2fb39b06e2b71083" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="657f37f0090d1d548ff2a41c1b4f8f6a54108b5730995af0c753683df9c81987" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="f965f9a2b018bb903e89ad1eeea3d2d8e6e702aab0fa000c5b14934d527d9a07" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="6eabbc4b0c3cad04ca36f8711b76e62d330d2b3b69845b2a47029b87757fafea" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="2df200dab3ba8f2fe4d9025236a3b7d57d8588013567daa16e8562f7a83baf9c" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="13e4e03439af712a56acf9e2b078c92c0bc2f7b39e14593ec36a28a4800639d8" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="9b06a1d20feaf231c34025f158eb0c0ecbf5b4c84a0c7d8c1b5ba648c8e50538" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="0c0759e1fd38f49f3b43125aef26ed8614d0b52dc8d12a383549070b115f13bf" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="d8580e810f8c611594dc8cca73abf82636a9c440548ed1ffb35cb75436cd6971" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="05723eef49882a9bd0be5bc6e7fdd65a1bc4d0f4cb52fad12176a50954750536" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="4cf4105303927303a3f055fbe09633f561820b90ea2a2e3aaf8f2ab0d639d697" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="e6b0911f67dc710778d75874698be3fddf33f4558ea2f5b9698215dd42c338b4" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="ed32384e2b24991be911d2812515ee64f6cb3a04a7bbaafea8645f323a8c5763" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="66cbc0cbf6020c4bee4ca9e88bccf56df8066b04f351d9d581f3622ef408edaf" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="658cc328411164ac0f4e8a7341930e489e0d8c957c07ac0b3b011333801e5b8c" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="a35ab42c098e4bfa9b2ae0b40016bb9adc3c93cdeb73d47cbba7b522543b9d7b" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="83f67c29cb2c5b4208c22a7e5090b0aa0f65aae0a60445ce367a31a1842a4ea0" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="f7ffa07393dc043e2cb9f76d69f26f4dc35626f9d32b3db922199886d196edb4" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="648cef2c45ca0cf88d9184c5e26407a420ec04dd674a20b468a6659608562ef7" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="5aa2dafa9d2ef2db72cfed5595326302c70dee21706bc7d179e61bfb8da7ab80" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="662af1e0ddfb0e64715b386f3b1ebc2d0386eb3901aac177c71c184104253e4a" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="9cb8fa327f9c0a6f95fcb95a321ab1e64a31c0c062eeb35c0085edc1ac39d508" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="eef26496ccae245331711efd8794eb6c6e8c7a6b7eb6b14776d37fea3d799a13" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="40ff526ce191ee756d6c2549f181776a1b0c85f2952b123594317233f8dcd649" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="1d14b72d1fa076d8f484654a1138b1cd5b726adfd0a1b6d9801cd68073a09ba4" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="757b0db024c8f6b95225498d39d9f65eb1c2523cf858c18cd31974eda3283726" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="48311576880419e720eb338f8b292e0581b923d5b670075e84061e0f5270bdb2" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="8b8e1f8f6d6c06f885a219a25130968472822b5ae98486cf63e1f37ea65d82ea" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="402d7a139a306a7373aa21cd6161732cf702aa816503ab0e970cb207eac78ef2" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="6db90a39ecc590ea854ecf9c3758db553f0db0188672bbba5d58f44a38e557cf" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="f899adda5e55badf0afbcd2b16fe3d5d5ed1535c245ec9401276742f725f2ed7" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="ff6427013d2e16426289f93fa324bf8f71a936011819aff9fd856bb26faf0fda" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="398159f3255a897bd9a84513e78fb48622520255052ebc89ff1eb360393c486f" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="7eb2d750b58519009655428e89efc0d49f793929ae9e2c7dcdd0d5d9eaf900d2" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="256ef9a8bb6f62f29a753e02b488b384668a6f3221d43c6357ffcfa0a6ed82f7" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="cdc3f9cd08e9f37909d3792574c929b27bcc2432bd0ace36759d95fd19ee96f9" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="8a296ef15c04add035c4bc37135a486f8539db1d680c5ef16784337a61bc3a00" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="de7b23220e236519cc219f6d5398ffdc1fb371d01a43ab646ab4eb41c6089eaf" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="a1d1b24befb291e94b2c5d24bc90024889d3aa5e0d70249cdff796ed82aee032" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="6809324402fa22ee0efa50a1e7541918a367c4be8cf5b72a7ca107c7b1a9f2f1" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="ebb0dfefb9d89604540a97f26022aa42c2ba6824b03a4ba644b4c2f9da0edd05" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="b232fe50de148d80a122790748ce3123dd93b0cf8aefec391ffe53799123fdf9" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="d351f474b74d1bcd931fa5e21ba258b44225d830e4e2205da0ff950a6e37ba9e" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="a3e239f362283d27935a10a9251869afaf30cf51fa395c718e233449623909cb" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="a5defdb8322e7503799f676f99e2841791de609a167f71377d81b8a0ba9b46c5" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="cf1d092d9fd675182089886d84628535c428f5bad7f7ce081da6a909c6efbb7a" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="a5be7c25d40bb3e4d9e204d9d52045c04c99d07873789b150de2fd1c493c72db" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="07b5b8fbd3ed0b02512a51461d30c1195434e4b4444a34dffdcff4fb5784f3df" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="2411736956ca061cfab43943877f000c5e3135f98e1da4690e91e6e9adb84f07" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="fed8da64ccc4dc4335cf9c0cb700da09ae12b92df0a063b4dd6ab4822d2cdfb3" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="ba79e403328f3cbd10f10803a3717cd87b27147c2216d11f0f0f97a59f567c24" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="a1daf1d9db8078326eec9ccefc6388df4f76794b903e01375cf37d25b5019976" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="0c5cb42423a077b582e33dbf231c056813b0a64bcd90df170fc11ccb21a45145" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="f0b9411b50ca1feff9360a0870fce5ad4b8271fa23f98fc71bdbee0c9b756398" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="c15bc7790983decb8ed7c60fe25b11fb8e1032b7a6c87ee088adee158df714ad" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="36caf99876bf3e3485eef2f1c07e1bc7f1f660967021cd16cd699f40c6ab7391" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="9bf78b545990c9365167899d24adbdd90b0b105c660c1b77be40a9dac54faeae" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="d6792ba10a5d2d4adabaf175640b100378e4665b81c92b9348f21cf0ed55284b" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="12e0174d286156173f3b5c17deac86cd92dbed245ac46a372fd57edbd6ab4bf4" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="db35615d5853e4943ef2d7250e5d07a834e36b7e23cf1999e1a5b43361d8d49a" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="8faf0948ad08fc98a44607256ca4be6f029d0d84bc0739c490bb49070888a6fa" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="92c77a33b5b5747fc489e494da8562c320eb97dbd654bdabc1cc00dd9111704b" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="6cf69d6bb3237532598045abb205def9dfae6ad543043300e8aedcfbfd9a4774" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="bf63476e687c147abd49ac54204ea2d460d5e2681e7278093d2dc2ddf29523c9" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="3afaee4dd16800975009af72ef3d1053e1aa123168f06c2489a81a3239f85784" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="420e5d172a3511ffed75df2aa413a3ff126604fcb70bdb1b49d2fffb1470cabd" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="95e1dddda6add5df42157d61da41b72fe23ada87e43763358fa2509e88084893" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="18df58bde05d421823da7320b35ff18511c3c4078ea0d7b5044213a015870ac0" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="dc51d947a4e03df8a07dbb1eeadf8f7ddffc3b577338db40e86221d23f6e4ca3" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="de03184b6ce31f434b508e5ba44a3d08b2b91adc989a4e95b30c593e7d87d403" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="5602ebc1b12951b64134a0c2990a9da37f5bd55f0289ec4870dade952e71fe6b" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="85d8289535eaf6ea92af3bcc0d7ff77272e6eb74791baa00d04addf0c21898ed" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="d2cb4ed29fb5d8c754cf43ee5e7309f831e6d4e2e68d70eade796347772f5030" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="49647339502c46db42e9d179a26d3d147e1214722fce59d48c62815c2f8e8b5f" name="asterisk-ael-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="92f8b527d05e9c6bcf8acbf6cf8f324a148903390e208013fa73a52f11f96a07" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="6af2291af0c7ac7987ffd2536242425a37c2f7fcad72c6acc817d82171b15f9c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="1b19d35f45d2992c6ff97cd76e3ef793ae3551196f3bcb6ff895d243ab84c1d7" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="5a68ca0aa6d7eeef03b705b2d9d3d4aad0b04d2d407c8ca5038a2525af808ffc" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="32416595f93552829862761e79a3683bbe3aeb522266d498daf4136c479a43e9" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="cd78fad7731fe63c957e37d762711cc1c2772c4e9c56520dd59ca7ad12a4a89c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="e00db74b11071076917d7907c5a4e2718efa7dc88472c84960774da4e95dfd6d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="25a2e8777da60c4819124d6c30a2636119caef64bf1555644d728b7efb1175f9" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="aef9daa1aa77a1cff8efd3d5bf0197830fcc6ac8d1dedfb789ebcfa5226e3284" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="d7f791038c38f03e9286f81a63027583d1aefa8c0bf1c74d28cfade45309b424" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="fc6a18a3012b907834f90dd0cefc0e62c8d0015cd06c16572ad26e8e343e2c30" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="0a84f8ee93c6c7daee0995b1cd28db236a9fe039cd81f901360f5e273fb2150c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="3608f124611b8bc4b13c9a4bb7bd5479da32c7b7eaaf399d412ef330f971a144" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="49c5cbf88e58bc70e78c63c0eb58b8d2e8c5dad3de2eb52b51ddef04d2d6e5dd" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="5e07f828f86d57456f31ae59fbbc55e1aa61f5383513401dfd5fdacf8baab71e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="a6b0913ef930f7af92c1edd331b773e18d8a9df790d302739d65386fc9abf0ce" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="a086a4bc43215fde65d4635243abb2e78475ac9a14ebf55891cddb1b800d080e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="b926e3452d2ebd4fd5ff342c262d6f1ac6ca128049e57ad8618c85db831f9ec7" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="5e0d9c8cc44e4c1f4a46799f6c12f5905e1a36daa264d527dc1f3446dd9e126b" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="0eec549d0700e69e25e873f350374452b2ac1aaed04204b4ed65c91695cf551d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="8afa46d24ac22223efbc457fac076d82be765318626aad9c06649a8537f251f4" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="d140b174e55bccde077b46fb389d6fd7d8608f82cae7d4e6ae328137a240a95c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="d82dd5e40d58b3bd772872e28e7da5397e89177c02222f1c07b0af2af6ba908c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="92fcf99dfde6b1f4dfac1af5f850a53523bc69b7b939f59c234a0be647b9b708" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="6493b8bf5965f342890f6f43d515a30625e731905b7674a2921fc03c2f62c2d1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="c10efac1045eace527c9c513fb3e332cf8dab6e4f34dcfac7ed853f102b431a6" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="5f50b4fb39d10364e09e9286886adefe64499ffb10f7aa33ecf4f44a291e17cc" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="97fd8cf681fab9a7548d8f97783af0d029a8da9c36d6337afee838813c297631" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="3d8bcc8bf7437e4f4d2bc499907f1f5e8848b4fda7813dcc18b5225ed696f853" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="6f07714ae6a9186a3e51273a782f0bd4e8dff11528a73ee1a2c880f8bbee1c03" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="81df395e4112acd2ba27ad12cf398acbd58c106fa1a003932df3d64b1b29c5b8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="f927005b8c75052958fe92053f7ef80d753bb337f3caf0a0a2476d43374ce682" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="70e3d2e305557765cc20190673aa322035e5976251432a21b9a584afc75cc95f" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="599b787aff9f441ab89ee87d73c2caf84e6a1344a85e299a759f44ff4bac4367" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="a350138bb2ffff7817b8d435239fdcd9528f3e3bfe133e0643e3573bde40feb8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="e011cd189ac6e56f396145112e4e2c3f969e46e36a36617bb38d81e242fd5580" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="20f53ac6a34af8440c1acb5ad2621606bb325307a10ec88632d74fa2d6597cad" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="5d4094140468bb391c072bcfc8d8a1e61fb6387cf2b1b17328dc11761d5a6a56" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="94bf4da8dd40851f3355c9c046acc2e264d7e88a994261bafd637e3a8950da17" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="8ca3eb3be3a1326d2aefebd6b1037f27bf1235ee5506c4fc1bc727f40242704c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="b5c64c3ac59020e23a37a21d543fba5bfc335ae254ac2d1d61efd78aec1bc279" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="d908c0aa354f5cd9540dbd49eab02f2c8d1032ea98f855e5fe63d508ea93ff50" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="a668238a67b2ae13f5b3fcd068d9c3aa3184fb6f337232e9f4a729389c9aa1ce" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="c6b1f25de2741597f598b994e4b4c69c19eeab48cdc1613d7325edce4a1fcb22" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="42682ad62f6dd2591529bc6c2681c2cf2f27be15e60da406f7a7c1d7669fc853" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="693f65ae07237d0456543efe629d7dfa7365797c21288e59a5ba201f329915a0" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="c1393582d1d3168282f4e21288f76aeaab43e68736237bddd604662aaeb559b2" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="3df6d8b1b0f5e442086b741872d73974b57d6fa9e333a6163d93363fa3619ace" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="29702e8123daf70e07184ca48d00a880b8a829890daecf3d73de1aef9ac9e6c4" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6f698a365ce71eb39f76dde1813eebfb916cc426a12a94b85c3f8b0e1021ef78" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="25259f9add58b0fa68d4675706957ecf7a0fca18b41309ce496964aae6fc0a39" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="f0dc5a7b9421d9c3015c705affe4f7e9ef8bfb4b2ddb86b5d40902b80a594a6d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="762e9d38ea54bde1edfbba2638bc724cad3b8ff2f75ab70b5d90926ee034742b" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="da2442f48d3e35f0f057da15bb84af4b44c18e2874a758fcbab9bae91d843e93" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="7ec5637aa94ca377b32f8e7e3e931602446219e269d424f214ded93736715c89" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="fc3830151e3a48a3b4017044504445c00ffb30f76e501ba694a523fd88a22492" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="a4760b5cf7e5f14bae743dc1eee476521020808cb05eb05c424fd85408c30246" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="dc101a647027164c571b19b744e6d9afa37914491a38175af8f0e55fd531e4a4" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="c762ad7da7f254b772291a16a586e01f7b561e9494a245a711b0076b1dcaff99" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="bdefeedd866acde4768a4b8c14e200212cf65f7e6e8bc803898436665906a023" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="2a2a80bf178182a227625a4734904b5be04f5b1f2e5600ba19a052d7dd2d2ae1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="73dba48be852b85c405dd94b35a9edccc0df7685945c3e07cce841037e36f618" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="7b12920a716f345c90fa7f5a3ffcdbbb16db6709ccd66c573e8be986970df50d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="b48565ae939282ba7e844bdbd121f74ab62255df1871eb2627ae50de13cc7996" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="ccb7d64d01379c76ab5192d9a8e72083039e81b50685c9c2ba571c97436c5109" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="00e3c9a904dad3576ca4c682bb1c7a04760014ed0801dc08d77c65c56b346d9f" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="b11007ed1d995781588ad23fb099bd0756236ba72659c398156d995cb638daa3" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="e2317414e2e0c8f146ea1daf3c9e5afb397eee6e08f68dd576cd36addcc5323c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="0758849b4c1041a06bc315d94d9d3e14469188227158ed61040951b80f965368" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="a9f0a8103eed4b21e89eb6e858c9c9faff19a5157fcc20b92186c2ac9077a850" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="1e095d096e0379f788d2d77e70d9075e49de7d1116d425a23af8575d465bd1fe" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="672a48eb97db2a92b23d409252b1be0d59fb9b0cee2217110c65092fbf7cc9f6" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="552ea4abc304e05b9acdfb5847309ba10d03a92e9886e39dd41d4df5303443d1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="ee536683b7e528a968d1d8adfdf073b0cdced8495bf6810bdab3a597a5a1bd54" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="7e3cfbca52c0dc30eca1c1c5a0a6decb1f8c226954b2a97b0c017aa07afeb8f2" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="a00a13ad9a2339abfad5edc4f8778f8b48ca84f4646166fd48c98da316e8bce1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="114978a73bcf7a66b23f8760730e7387bb28091513baa5c6b1d3afbf29f74bcf" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="273b691661d3e0ea5d77fc782d364241d73ad1c7fc6ecdbc41c7a2985572415e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="9746645acba13533e1d09d5986cf45302498ff88f3ff79f4de9defea640e898d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="c943f24ea5ae4134539d25fcc7a4e391766f7a1745c43d5765ed122f949a09f8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="1959eeaf6207dfc77b4283423429a62f9385d8549003045c2b5e3f917b675bf2" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="adc57278d729b44acee3a70b7ad25710bb781bf8c7bae4d3c2b28ae22484bb8d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="17527c109a5fed3811e9fbd62b9c7854ae1f945b00a8780de6ebee725c519cf0" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="9135c759282efc254ebeb43649030c3f312b2edc8f628abe3c6fc6bb244fd79f" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="0badae72c7cc5252ca5af0adf4b7b1655c4c66a64b76860b7f610aae84c7233c" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="5abb3e3f33f3c03957e113ad097753e4eb2469b0c7446f6f1415e584f37fceb7" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="9015ea9199dc54856aa74086e4d8b1f64009a4840e79a0465deadf0ad65d2630" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="6e78a71164758ec0e2b1c37f207d26f757a753ff22f5c64737a4a5b33665219a" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="6a92f7b8bda52ed6c948851f1eba39a64a42b7b862cf81a35b0dd00af1d189b3" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="52c6c1251c333ccc6774843c1bb586a8390498be4038b8cdf9af8a841c691caf" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="93f4d3de1c67d25ee406748eb7ea6e0947fcb96617d73cc141daf77eeeef020e" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="55b90a853cf5e524b760e87ed6696ec843a1ce13ac2373706162d06dce137eef" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="2d4c16a8543883d258b985b60494b4efbc3cd0665eb3dd0a0b006bf720a2f4ef" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="cf031bac04acc3d62a020956daab528a923c551f105ca04ec19b8d95e2eb27f2" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="afb3e75de8ec7be9a47ec52c07c708be131986268c61c97ac35de2c46201b628" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="606773666b7c98412234339c65fadb9e0a6c9c27e084e3ec8c71b245470ff1f3" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="fd0dadbd27cd57a96f7358c791420c085e1cd9554374cc7a3283d3f55b20ceb6" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="eb52a481267a0c74eb14f751ead4deed82e481b3a44cd0148f23ef774a007d0d" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="eb685ecedf840ab5c61a0c28960c38001b891b7d281bf543cdf1406b46de8e61" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="186983bef108c236a3754a7ea17eab1945fe7fb008fa2c57ca11e73b426deb2b" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="fc756ce16e786232894e93df0ec9b7622411371a40158a0acc3284ab80ab8488" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="cbdb4e1f13071afa1667e58c04359f0a1b64b23364579eaf9b20bcecbbff8169" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="1dacdaca4993ae145261440d7c1342eb18d595dff320112e0b5d6b46b1742b14" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="eca16d4f6ad420c4200dee22364849e4b1149d4a4779dda1715a65ebb0a4f9cb" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="837e38f1ef46c0a900a60455777c6368aa0e9634808084e5009929b7614851de" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="1413199b206641fcf3843f5d6b17bcd82b11b97c955804b268cb3703cf65c84b" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="0844132692635deda49bea063827738d9161bfbae893b81ed6cdb8d95dced549" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="38deaae87fa5f490001b0cd6d758d6861e04e0e27de4f663b1d24c49c076f7da" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="2d94a1e56076a69ef035d820544069bb79bd157084e787340cd094246d380075" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="4f6a190293e55c8fc3c0fc6668cacee687fb7b76d26e786ae67038e669b9733d" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="bd11e7e210f575a48bb7ffd8322de3529bcd547c7d61fdd2d87970a4ec2bfc07" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="1d7353b4887cf7f6a264f1862c3fc84b97119ae287062aa8ffd4ba99a03a5665" name="asterisk-alsa" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="cdb43a04c474a4c60fffd199c7453ef47bb3a0588835fe1fb2b441ad2d516dfc" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="1e3a3855d12cf67dd015022f7d2a67d7c773de3e675b280130496abc9d37381e" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="138c4cac50873bdbc969a0a84b7bdb5a6968b690cefd2ba44745b34b682f5681" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="bed0d37202d526e69ae4b152ffad37829f360f35caca2d675d65ac110129b7cd" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="60b63cc29b31b1d5c0fe29077c4fd06826916f37e2bf2e45dbb7c65938be45f0" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="e4be21437a7b1f6fa9a81613d2d5b9f25b627982dc8af7a9430e2d6cadb92899" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="a17184f0f84e9555538ebf16228435f4d70d859dd0b8eb9aeb97a391d88c6b81" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="e0ac927463040b78d32b3eb3932477b85adbad930b819560b7804cd86814f440" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="575af9181d3cd519c28d6632ed02aff404cb14d04211f6bb4b10723ea4050386" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="9ae303ce815ca2c4f068691da5f93e003386eb5aee937f9a8147b22b85c4b93f" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="9099ec102b0ecb9f05108997ac93fbcb262171c86683b164d04d5207d5f6acec" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="41d0dd74a207c03274fba79bbe41c59635145ec77f9e36866d98cb54faeb2ec6" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="3016ca5d416e96566b688b923183f284c38a05f3e70f5e74d43818807a4a8d07" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="4279127c894604836b773713a711891873854fdaa119454989e8cb3193818154" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="edd68feac5d2aecd99fe85b299c878fa3e36b8bef394960555c69a9990b7a283" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="c01150630cc41ad3c104e0ec0ff644739234071206a82a27133eb632fc595bfd" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="6aa7347c19e555db0a73a6cf38b9448da8b2660e457e5b64d27493320d97959f" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="755524db336f210a9c29588d657d9d744a65ef075c9dc6884c137640c4921d34" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="57ea76d6e918d50866b288b477d9731cf80dde9a2527c58585a37ada6d24e8a4" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="f1b013dfc77fa70b6b0389516e7c4f0383121b0a44e04d53dabcbe6b4b81aac6" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="2c86bca3f085f538b58266b7937db17f8af25fdfe9597ad17c56cb5d08c23388" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="dbc7d3bc7dbb4d94d09db3d661e3adb07fc75172223f8d65f2501771ae709131" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="5117111b8eb89aeb25d26276e774de74d79f507e0ff4be82af1b815948d218b3" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="7fe9ea9c7483657da9323b7dddd11313022b3432f741b35dc35b078cf6ed8c1f" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="47f457b19dfc5a936b0cd70ebcd7a643ca73ebe950226d275d2984eb2b5b184d" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="f2827379f068d57190b5ec3645b45506706e74ca22affcf6f8a18ff607ce07e7" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="676988552367a782c40a1501445eeabc37883842a76d69bf6fec758563ff29f6" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="e0cd9059417ad684882abc9298e95f15dcb86aac274bafc3710c0edd203015ef" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="1fade060bd4813d2057b0e80c26323dff65fbfa9c795a97dd9b9f354aa47cc28" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="59ba0a43acd5e0839ad8f6448cbe11c058a170603fa4f29790a0f70cdb5e13f9" name="asterisk-alsa-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="48b1ea4252d00e43b196974d8c8c5e9195ae86e9107953ce86ef341a8e5bfbeb" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="aa462e850bae535110acdeb558a518cf82ae6bb654e7b185dfb1fd342e04f66b" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="55791d7dc06a11e309856c5a039a30ac735551e4da85faa65a6dabb2fbfa6fed" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="428d307155dc1cba4cb629abdf7404498d68e2d302adce2125ac6c9774dc7d87" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="31bf08a4053f00e1bf05ce18c656ab7c7d9fd9cf45f9924810d293edba859659" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="1738fdc28a5406e21a2c8846918c3f2f8ffa53c51d0c1c85569cd78c7e61be8b" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="d03423908c61a334591657c9b95db0a8d429c5fb76cc0ae34dfa51c256e8152f" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="33d9d7500c7e29d28e2ce957104756bc18c012286b95e1c1d87dc7338ae1ca4d" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="81d7c8b9e245e4b6aa58e5a551b66ea32098355d72b0c673074c617c67270a33" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="c0f23178e8ec1c56edc8ddf48fe9276d2a34075a2297c67e9207cc43ff31c7cb" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="e90ff5fd2df1619f7046079c451f87cc4d3113c8563b7916b5a956475075cae2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="4f6355b995ec8bf9f62e401e4436388b6df8cc798c49ace9ca2c7cc8205d0d20" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="dcf34314377c08d9786dc05ab11bda1e69eeea20c83ce7155fb7be7cf522e968" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="c031d6895f17f4e6d652d2105ae1b6facfebffc65bd7cef4c5e19fdc67b1c6be" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="964e932af3179465dc1ccfe459cb4f1d30c0585edd3fbe00f90bc381586fb665" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="5cb5cddd4c5b24592f39469b15e431567f503caf5e3d638df9b3b6805449c9a8" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="c148915412a4545e73d44fda13cf1df00747c83fadae731d0051caaef1d3e0a5" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="bc2c58efd05e61763def70de1e5d8bf0b8bdb65809428092b4f5a11548d8b616" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="5b5188dae8a38d59029bc2b9248837fa488f18ae848ab303249ff2af8d5210be" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="d8426c15edd6b4eedcb894cbff151a80e5f477b8df14a77e3bb0ce94d0eac1aa" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="be4424709e25b633bb88e7de3831a389349e3cb312bfda277d5dd603ce1193fe" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="aecea6c9a9c0bd2244ee309d4dbbfb24cf1ac4a2fa10f73c6f0bfad81ca284c5" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="1f1a83a88e29ba689ac131db844d4afe46a693fe3e2dc6850eaa7b51d1468bf2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="4f6fe3e9b47338768e3bdfdb6aa7bfca80ce7842235f9b3699f762d041aa8239" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="6d48d1a34c50c12a1dc9e31b80b9ec6b1e48180f4c149f51e8c4103e096e59a2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="551172f26b1af20306f6d8933bb6d2af34ee6d8ea9ae4681dfb257b15bfa30fe" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="7adbee26932a7d88146420fd8f8beae91d8c167bd90fc1e2b2e4c8d90af95856" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="60050a71cc5db4bc8e07012cbb46fec0efe0f9a27ed5c0660c0e1df0809c703a" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="3805f567fceefd2a1210e16f7d84d0fae8e6ad4c0131484b92ca00b9c9e154ea" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="4d10030dadec7d271d85f971aec5c2f55f47a5287c88bbc01d69b78bfbbb842d" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="88fa0761523be070d4fa290ee9853d37237d5c317531f6ec5da0ed32ae6bc778" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="1b34d791bb006bedbe30d4eb7bdf70bf61858d856718c66ceef8868341d4c5ab" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="b1ab0b19945fd9678f01c3c12f834545a5e840e1059d846623eb574bf868b234" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="ddace20203659d277b4506977e6e21087e0852b664f70692fca7a9bb0c7705f3" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="2204fc34a303fa9cfb903adcad8448bf28e72b0df7b83c989300e8facda94d3a" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="d0e30f524cb1f45afeab1febb37183523c7617b7c2d144ddda0a3dc6123f6e4e" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="fc2441daa9c19fe2cb5be7a2a759e90f01cf63d2990d5f7d7869e9b2daa249f6" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="f4abeae8d7fcfc7b465ea5e7f4661a17e447fa9bc05d6ba2574af981cae2c315" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="e882b76cc3277740b244b7bfd3c212d94fb23b23e19b3b0ccbda332a56352e89" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="38fafaf0bca6d47dcc842fbb53caef42717e9bc70066287e13f9adbb0a2d8a53" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="43ba6e38acbc1c0618c6d91932a7c12811d18fd7cf0a32d0f806648929ab7182" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="e08d1a2b4c26d000d4dd4448d2451af716f20f0c607777509e651f9bb6059b93" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="c32fa6063712a4a4f4d8e0c7fc3d998af2fe395a31a7fc2ff509b65edbd393a2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="0199fd38284d63fe7e0eb1e6f0f50ca4e1862a6efd01ecbc551c5421b25edb74" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="0e14fdb7182f374e24b37bccb9dae2dfaa44664b124db526d794af6f35a1a8d6" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="7afa8c5cf3c0dd722997096934c582e66acc0497405052a56e53621a613aacaa" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="6ca810cd18281bdb8f2ebdf41c63992b53d879aa92a05306bdf2106a7ad5da6f" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="6c4b1a1b2645beec16b8b3d8cb8a5578aee0763cb501de111c51777018d8e220" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="9e8430da48310dd215e8f276ca42652f977032d29114cbf238f028d6420d7928" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="37f776f978a9c0e2ab394289d0e9333e280cc562f236b538bd5d234b95184120" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="02eac2f9dfde5bc8150c69a0053c65a01b0341a1c27aeb657d607842f7f941b1" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="a74ee418b1a3ea7b6f1b439feadf12a4c48df06c5bdb0915e20d935ffe59f453" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="6b679de10a6c34f96779fdc781ce1b20d1cc823c58e7780ca55c063e7521648a" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="5241efee9f796d9fbfed3dbfa3b6230e1f0ca36083a1dbb631dd163218f485ae" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="619aca251ebb5d7d4f6a7fd795226e809aa34cecd57b8c70c62497f172de56f5" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="b4f2a198ec7d5611a8e974d7d835caf44d9de9204f8ebcba721245b9b238e44d" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="fe4b0ff390a0a5da5d40ccb3ea171b89c29f23650f39f8e34d921abbdad0b08f" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="36c3820581538f96c96b9bbac7a3def8791b2723b1a0311d028a1f35a81fa46b" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="759c6262212ec44c8d47f2b3003a98918e605c0bd36cc15ef82418f76e8d3b77" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="7daa253af45f4c7a2c7c388b183c9ad3664c3c23ab956671d9bcfabb4515969d" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9bd00abb1b9a7b905ac253b6db241e19f2bbf6bad8f64b3dcfbac5af6aa20cf3" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="8b8ebd1e517358ba1e2aed800aa492177c7273265acd89e939c07b25df9c20a2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="16b548a149a3f450a1bb182a84e7f92994be7787dc7317f9265c53cef26ed8b2" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="3fe7a4234051a6a03017abc0de7ab08d7642cb3a34e7065b0d0b351f02205bed" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="c10a6426e6013a2bfd3b7eb5644d78801cd7948750de91502cd967e95329443d" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="186f6d39d0731e865ea769b18073349d03862f6b351c9e3b9fa9532b5bb9606e" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="7000df3fdaaa55c108236ae5da7511d03d334771cdd5d2bec05e2ade781735f0" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="2afbba08e03ecadc7a5d875a17358a7ef6477c4290a350a94951a86ec26e527c" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="0828d844c598ea85dacfefbe08a74556b2ef6520d7fa8f542aa1c18b169b6d08" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="59a8deae23333435e026b5e982caa7a196b3afaeb4fd1e23cdc589b672b81b6b" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="1d1fd42314937e11f069697823a1cb3b932ea8d220f3b872e3fa589c3d3b6c4a" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="d910cbee441426cf9aed3fceb3c67650e76cdc330f8dbe4bd1f1375e32e23bf3" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="7d42acffa3af028fdf051146729fac76a61fa15ef14e52f187d232d07bf79bf0" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="4fbd77e396ffb91bcb09ec72523d98c52ff5e90eff377bbdddb05ba5077a6e31" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="7822ab7bce6e875856908f66cdb49c1043b3a60c57671574bd3fd6bdd52b3aad" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="549e3a6a89ce4d421698bbd8aa7db9719a5f4e5be4dbf21d297552374bb8e0b7" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="25fb7bd9d87e19175300e8d0ae861588714ee17635848aa292281a5fc4b85b33" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="e75e6c412272a5966797e964085482de194a54b12c5366b3dd5129c4a8e0e96c" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="9555f13c079babaa148247e1a7e9f35cc267905687c093de7c52034efe6631c8" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="71a52530e11e8eebfb399e9fafcdbbf366fd48aadd7fb422ce53f03c76091947" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="f5ba3f3bbb9cc2599ee55de221efd0e65bab9efb5ae9092f5dffaa47c89e099c" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="055af33a708308f7a66722fb2435c18db0a0780b085021510481d0aa356a87be" name="asterisk-app_audiofork" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="c5a6380b823117af94cc72b2bf7e0b6d97bb22543b15bd64b53dda857756d849" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="39d3777a66b55166ed6d3dc5844e4f2fcf6e576808d92b3dc91fb6f37df31c98" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="88bb757bab4fd9fe66e389d41716a0a36217e61bda05f753f3b9b7018d94d468" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="66b3750852043661a0e6d0d9c2620840dba2984c46f886e12bd022d580d6a0a8" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="13b517ca2dfe3eec4c3739767ed47e94d5f878760471566591e6327c0eb84862" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="d817baed3d9b32105affdc64d5c53cc77d8f4729c0688e0accd9724c6654ae74" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="ba3a5b895667a3d179c096951338aa7b1de03e3eed39a80c117d17a4f45a276d" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="8e4ae62a4cc4987847766033391140ddf2be0217569c3622d6e0c2dbaeb8762b" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="7474a076bb85038164a5d38170eaaa696f9cdc3732d53fa366e2063d32814baf" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="fa9d2882d61111b59e225bcfd5b448dfe89027f745c1c2a6b44660db3e447a18" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="f2232a50e9f338589ced7a2741874692bcade07729bb672b4db1b98ab3744f0d" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="5dc9d24874b4b3de3b294a6a71e32cda4c8770c004dc25f30dcb9b5958d2f077" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="1b0a3cea6e2d114b8b82c7e9a8b98baaf1b13d26505f83944f93c94b0bd2f13d" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="794fa5f739b0aff95e5be61b49c14231aff313dabe24d264fa4fe1c1d1646239" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="0632ee76c4bbc704988f121333574f0240338afe31b411a62d1eff17a418d9dc" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="087c4716dec5fac83ad2324b4459b469fd8b809624c7d77ce2980530c3a382fc" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of 
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been 
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="d5e540681867c3aa2456c5cf4df915bb774c3b59fe471c2dff498cb49a2adcfb" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="43abc7de6180f3f243149ee55a9eccfacd1f884a09a4c0309838d23e7159963c" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="dfedcc80af455ae29f74aa28155169e9c1c9a320a5bace6495a4f473cea666d7" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="919371ad78af68da946c2c388e2f231bb0ecb3cbd47de8301dadd6a31122467d" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="8f97e4cc4cb1c8eaa2067f6bc742a5642a1567608dd4d7f23732950ee574e3f9" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="3ddb410fccca0ebf0c94fcb37ced33600af9a1af4b98d5a38c7815f8203fef92" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="52d91e38cd64165ac17e3029930e95754443c517d873e7748fd948bc67a52e50" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="c3cc82d4b96e278e12c44634279fabed3028116acc896eaf09f4799c7fcdf1b5" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="b71059d2a5d620edea22d13e66974e7194e2c69d3d517f9eae43de3694157307" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="f6786b6642f027c010fbcef8ba2fc879a88394355d13e5765edcbc84430a86e4" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="5c86ad5ba1263c438115ae4f63a75ab6e54bb71d01d3f9eaf25d9db80eeeea26" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="32d36b9547e03eeda4653765ef2b7e8dfa598fe1ce78641f0e11accd89bb592d" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="b9cbcb2368ea4fa409c31ca9f9bb1c20967dfea196c61a2aa2c708aee734620b" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="d1bac15ae04f783430f4ea78fa723e81946938baa7c7f619debe3467a6703f97" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="d824192fe296db5fa5a80ce43abbd8ba567ed2e939a2ae30db0a097c8faaf7f8" name="asterisk-app_audiofork-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el10"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RL